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Unified Diff: webrtc/api/peerconnectioninterface_unittest.cc

Issue 1837393002: Re-enabling tests that were disabled for Windows debug builds. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
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Index: webrtc/api/peerconnectioninterface_unittest.cc
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index 2c044e20806d1bcf0115f5b263c9640a616e7712..61eb5f0423c4f9a9114eb4140a3ebaaa0ab89155 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -1701,16 +1701,10 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_ReceiveFireFoxOffer DISABLED_ReceiveFireFoxOffer
-#else
-#define MAYBE_ReceiveFireFoxOffer ReceiveFireFoxOffer
-#endif
// Test that we can create a session description from an SDP string from
// FireFox, use it as a remote session description, generate an answer and use
// the answer as a local description.
-TEST_F(PeerConnectionInterfaceTest, MAYBE_ReceiveFireFoxOffer) {
+TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
@@ -2094,19 +2088,11 @@ TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SdpWithoutMsidCreatesDefaultStream \
- DISABLED_SdpWithoutMsidCreatesDefaultStream
-#else
-#define MAYBE_SdpWithoutMsidCreatesDefaultStream \
- SdpWithoutMsidCreatesDefaultStream
-#endif
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and no MSID support.
// It also tests that the default stream is updated if a video m-line is added
// in a subsequent session description.
-TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithoutMsidCreatesDefaultStream) {
+TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2132,18 +2118,10 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithoutMsidCreatesDefaultStream) {
remote_stream->GetVideoTracks()[0]->state());
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \
- DISABLED_SendOnlySdpWithoutMsidCreatesDefaultStream
-#else
-#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \
- SendOnlySdpWithoutMsidCreatesDefaultStream
-#endif
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and media direction is send only.
TEST_F(PeerConnectionInterfaceTest,
- MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream) {
+ SendOnlySdpWithoutMsidCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2175,19 +2153,11 @@ TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) {
// No crash is a pass.
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \
- DISABLED_SdpWithoutMsidAndStreamsCreatesDefaultStream
-#else
-#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \
- SdpWithoutMsidAndStreamsCreatesDefaultStream
-#endif
// This tests that a default MediaStream is created if the remote session
// description doesn't contain any streams and don't contain an indication if
// MSID is supported.
TEST_F(PeerConnectionInterfaceTest,
- MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream) {
+ SdpWithoutMsidAndStreamsCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2200,17 +2170,9 @@ TEST_F(PeerConnectionInterfaceTest,
EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SdpWithMsidDontCreatesDefaultStream \
- DISABLED_SdpWithMsidDontCreatesDefaultStream
-#else
-#define MAYBE_SdpWithMsidDontCreatesDefaultStream \
- SdpWithMsidDontCreatesDefaultStream
-#endif
// This tests that a default MediaStream is not created if the remote session
// description doesn't contain any streams but does support MSID.
-TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithMsidDontCreatesDefaultStream) {
+TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2219,19 +2181,11 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithMsidDontCreatesDefaultStream) {
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_DefaultTracksNotDestroyedAndRecreated \
- DISABLED_DefaultTracksNotDestroyedAndRecreated
-#else
-#define MAYBE_DefaultTracksNotDestroyedAndRecreated \
- DefaultTracksNotDestroyedAndRecreated
-#endif
// This tests that when setting a new description, the old default tracks are
// not destroyed and recreated.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
TEST_F(PeerConnectionInterfaceTest,
- MAYBE_DefaultTracksNotDestroyedAndRecreated) {
+ DefaultTracksNotDestroyedAndRecreated) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2266,17 +2220,11 @@ TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_LocalDescriptionChanged DISABLED_LocalDescriptionChanged
-#else
-#define MAYBE_LocalDescriptionChanged LocalDescriptionChanged
-#endif
// This tests that an RtpSender is created when the local description is set
// after adding a local stream.
// TODO(deadbeef): This test and the one below it need to be updated when
// an RtpSender's lifetime isn't determined by when a local description is set.
-TEST_F(PeerConnectionInterfaceTest, MAYBE_LocalDescriptionChanged) {
+TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2312,18 +2260,10 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_LocalDescriptionChanged) {
EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1]));
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \
- DISABLED_AddLocalStreamAfterLocalDescriptionChanged
-#else
-#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \
- AddLocalStreamAfterLocalDescriptionChanged
-#endif
// This tests that an RtpSender is created when the local description is set
// before adding a local stream.
TEST_F(PeerConnectionInterfaceTest,
- MAYBE_AddLocalStreamAfterLocalDescriptionChanged) {
+ AddLocalStreamAfterLocalDescriptionChanged) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2349,18 +2289,10 @@ TEST_F(PeerConnectionInterfaceTest,
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \
- DISABLED_ChangeSsrcOnTrackInLocalSessionDescription
-#else
-#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \
- ChangeSsrcOnTrackInLocalSessionDescription
-#endif
// This tests that the expected behavior occurs if the SSRC on a local track is
// changed when SetLocalDescription is called.
TEST_F(PeerConnectionInterfaceTest,
- MAYBE_ChangeSsrcOnTrackInLocalSessionDescription) {
+ ChangeSsrcOnTrackInLocalSessionDescription) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2404,18 +2336,10 @@ TEST_F(PeerConnectionInterfaceTest,
// changed.
}
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SignalSameTracksInSeparateMediaStream \
- DISABLED_SignalSameTracksInSeparateMediaStream
-#else
-#define MAYBE_SignalSameTracksInSeparateMediaStream \
- SignalSameTracksInSeparateMediaStream
-#endif
// This tests that the expected behavior occurs if a new session description is
// set with the same tracks, but on a different MediaStream.
TEST_F(PeerConnectionInterfaceTest,
- MAYBE_SignalSameTracksInSeparateMediaStream) {
+ SignalSameTracksInSeparateMediaStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
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